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2008-11-14Fix some refcounting in app_queue.c and change theMark Michelson
hashing used by app_queue.c to be case-insensitive. This is accomplished by adding a new case-insensitive hashing function. This was necessary to prevent bad refcount errors (and potential crashes) which would occur due to the fact that queues were initially read from the config file in a case-sensitive manner. Then, when a user issued a CLI command or manager action, we allowed for case-insensitive input and used that input to directly try to find the queue in the hash table. The result was either that we could not find a queue that was input or worse, we would end up hashing to a completely bogus value based on the input. This commit resolves the problem presented in issue #13703. However, that issue was reported against 1.6.0. Since this fix introduces a behavior change, I am electing to not place this same fix in to the 1.6.0 or 1.6.1 branches, and instead will opt for a change which does not change behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-13Remove trailing whitespacesEliel C. Sardanons
using ':%s/\s\+$//' pointed by seanbright on #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12This commit does two things:Michiel van Baak
- Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12Implement AGI XML documentation parsing functions.Eliel C. Sardanons
A new <agi> element is used to describe the XML documentation. We have the usual synopsis,syntax,description and seealso for AGI commands. The CLI 'agi show commands' command was changed to show all the documentation se ctions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-11use some fancy compiler magic (thanks to Matthew Woehlke on the gcc-help ↵Kevin P. Fleming
mailing list) to restore type-safety to S_OR by going back to a macro, but preserve the side-effect-safe usage of the macro arguments git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-10Move all the XML documentation API from pbx.c to xmldoc.c.Eliel C. Sardanons
Export the XML documentation API: ast_xmldoc_build_synopsis() ast_xmldoc_build_syntax() ast_xmldoc_build_description() ast_xmldoc_build_seealso() ast_xmldoc_build_arguments() ast_xmldoc_printable() ast_xmldoc_load_documentation() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-09In order to move away from nested function use, some changes to the recently ↵Sean Bright
introduced ast_channel_search_locked need to be made. Specifically, the caller needs to be able to pass arbitrary data which in turn is passed to the callback. This patch addresses all of the nested functions currently in asterisk trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-09Merged revisions 155553 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-08 - Check for failure when putting the packet in the ast_strRussell Bryant
- fix a spelling error in a header file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-07Add ability to pass arbitrary data to the ao2_callback_fn (called fromSean Bright
ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either of these mandates that the passed 'arg' is a hashable object, making searching for an ao2 object based on outside criteria difficult. Reviewed by Russell and Mark M. via ReviewBoard: http://reviewboard.digium.com/r/36/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-07Convert open-coded linked list in indications to the AST_LIST_* macros. ThisSean Bright
cleans the code up some and should make it more maintainable as time goes on. Reviewed by Russell, Kevin, Mark M., and Tilghman via ReviewBoard: http://reviewboard.digium.com/r/34/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-07Clarify which part of OBJ_MULTIPLE is not implemented, and under what case itRussell Bryant
is perfectly fine to use. (Inspired by a question I received about my last commit.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-06make S_OR and S_COR safe to use even if the parameters are function calls or ↵Kevin P. Fleming
have side effects. it still bothers me that these are called S_OR and not something like ast_string_or, but that's water over the bridge git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-05Fix a problem found while building res_snmp.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-05Add LISTFILTER dialplan function, along with supporting documentation. SeeTilghman Lesher
documentation for more information on how to use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-05Make compilation of chan_dahdi so that it does not require the new ↵Matthew Fredrickson
pri_progress_with_cause function to have libpri support work. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04Introduce a new API call ast_channel_search_locked, which iterates through theSean Bright
channel list calling a caller-defined callback. The callback returns non-zero if a match is found. This should speed up some of the code that I committed earlier today in chan_sip (which is also updated by this commit). Reviewed by russellb and kpfleming via ReviewBoard: http://reviewboard.digium.com/r/28/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04Slightly optimize ast_devstate_str and rename global functions devstate2str ↵Tilghman Lesher
and config_text_file_save to have an ast_ prefix git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02instead of trying to forcibly load res_agi when app_stack is loaded (even if ↵Kevin P. Fleming
the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02Merged revisions 153651 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008) | 2 lines features.h depends on linkedlists.h, so include it ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01Merge changes from team/group/appdocsxmlRussell Bryant
This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-31* Fixed timeout logic in the dialing API as setting timeoutsMark Michelson
had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-31Recent CDR fixes moved execution of the 'h' exten into the bridging code, so ↵Terry Wilson
variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-30Add a todo for a new timing API implementation that would work for Linux systemsRussell Bryant
as of kernel 2.6.25 and glibc 2.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-30Fix a bug in AST_SCHED_REPLACE_UNREF(). The reference count of the objectRussell Bryant
_must_ be increased before creating the scheduler entry. Otherwise, you create a race condition where the reference count may hit zero and the object can disappear out from under you. This could also would have incorrectly decreased the reference count in the case that the scheduler add failed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-30try to get this committed before the buildbot complains about a broken treeKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29Merged revisions 152535 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-19cleaup of the TCP/TLS socket API:Kevin P. Fleming
1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17Merge codec_consistency branch. This should make sample usage much happier.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17Fix the FRACK! warnings in chan_iax2 when POKE/LAGRQ packets are not answered.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14Merged revisions 149204 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14Add additional memory debugging to several core APIs, and fix several memoryTilghman Lesher
leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14Merge realtime_update2 branch, which adds a new realtime API call namedTilghman Lesher
'update2', which permits updates which match across multiple columns, instead of requiring all tables to have a single unique identifier. All of the other API calls with the exception of 'update' already had the ability to match on multiple fields, so it was a missing and very desireable feature that an API call implementing an update should have this, too. This does not change any outward performance of Asterisk, but it should make life easier for application developers who use the RealTime framework. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-10Don't include logger.h in asterisk.h by default as it is causing problems ↵Sean Bright
building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09only include this for OpenBSD. At least FreeBSD is borked when including itMichiel van Baak
(closes issue #13649) Reported by: ys git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09(closes issue #13557)Steve Murphy
Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-07Allow people to select the old console behavior of white text on a blackTilghman Lesher
background, by using the startup flag '-B'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06Update documentation; AST_THREADSTORAGE() in trunk only takes a singleTilghman Lesher
argument. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06All ODBC parts can now use either unixodbc or iodbc.Michiel van Baak
This allows for the ODBC parts to work on OpenBSD as well. 99.99% of the work is done by seanbright (bow, bow) and I actually did nothing but test and yell at him that it still didn't work :) Thanks for helping out ! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06Similar to r143204, masquerade the channel in the case of Park being called ↵Jeff Peeler
from AGI. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06Mvanbaak said this was needed to compile on OpenBSD, so put it in the ↵Jeff Peeler
OpenBSD section. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06make aescrypt.c compile on OpenBSD againMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-01Add schedule extensions to app_meetme. In addition, the reporter found aTilghman Lesher
problem within strptime(3), which we are correcting here with ast_strptime(). (closes issue #11040) Reported by: DEA Patches: 20080910__bug11040.diff.txt uploaded by Corydon76 (license 14) Tested by: DEA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-27fix bugs caused by r144949 when MALLOC_DEBUG is definedKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-27Merged revisions 144924-144925 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines improve header inclusion process in a few small ways: - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled - simplify the usage of some of these headers in the AEL-related stuff in the utils directory ........ r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines fix some minor issues with rev 144924 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-25I added a little verbage to hashtab for the hashtab_destroy func.Steve Murphy
It was pretty sparsely documented. This update fleshes out the pbx_lua module, to add the switch statements to the extensions in the extensions.lua file, as well as removing them when the module is unloaded. Many thanks to Matt Nicholson for his fine contribution! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingTilghman Lesher
when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12Merged revisions 142675 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-09Minor fix to docoBradley Latus
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-26(closes issue #13366)Steve Murphy
Reported by: erousseau This was a reasonable enhancement request, which was easy to implement. Since it's an enhancement, it could only be applied to trunk. Basically, for accounting where "initiated" seconds are billed for, if the microseconds field on the end time is greater than the microseconds field for the answer time, add one second to the billsec field. The implementation was requested by erousseau, and I've implemented it as requested. I've updated the CHANGES, the cdr.conf.sample, and the .h files accordingly, to accept and set a flag for the corresponding new option. cdr.c adds in the extra second based on the usec fields if the option is set. Tested, seems to be working fine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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